balaji wrote:
> Hai,
>
> I think I did not put my question clearly. Aim of this exercise is
> resampling the signal. I can do this in time domain, by adding zeros
> followed by filterin...
Rob Hutchinson wrote:
> What is the preferred method for removing 60 hz hum from a signal without
> wiping out signal info around 60 hz? A 60 hz notch filter would not be
> useful because it ...
philblack wrote:
> Anyone,
>
> What is the best method to convert a analog filter design to a FIR
> filter design? Lets say I have the tranfer function in the S domain
> of a analog filte...
Hi,
Has any of you already successfully written a bootloader for the 6711
dsk? I'm trying to find out what my options are.
At the end I need an interrupt vector table at 0x00 and a lot of fast
i...
Steve wrote:
> Hi,
>
> I am designing a very speical filter, really it is very unique. I
> don't know why they need this filter but I need to design it.
>
> It is basically a bandpass ...
David Reid wrote:
> How can this be done? My application is converting the sample rate of
> 44.1kHz wave files to sample rate X (eg change SR by factor of 0.75, or 1.3)
> and playing back at ...
phuture_project wrote:
> Hi,
>
> As i've got problems using FFT or Goertzel algorithm (for those who
> are interested by the reason see my previous post called "some
> goertzel questions"),...
Hi Mark,
>
> 1. My IIR version of a continuous-time filter does not seem to behave
> as it's Bode plot says it should. Specifically I see a large DC offset
> on the output of the filter, bu...
schorsche wrote:
> Hiya,
>
> I've been struggling quite a while now with the following and couldn't
> find
> an answer, please help:
> All I wanna do is read the sample values of a .w...
Dan Shorb wrote:
> > In the paper, he claims the inverse filter of the log-swept sine for
> > deconvolving the speaker's output is simply the time-reversed input
> > with an exponentially-decrea...